Distortion compensation in an acoustic echo canceler

ABSTRACT

An audio communications system has an acoustic echo cancellation (AEC) module. The AEC module receives a digital signal sent to a loudspeaker and a digital signal received from a microphone. The signal received from the microphone contains an echo of the signal played through the loudspeaker. The loudspeaker signal is processed by an audio generation module (AGM) that models substantially nonlinear distortions that can occur while producing the signal played through the loudspeaker. The AGM includes a modeling path comprised of one or more distortion modules. Each distortion module receives digital samples as input, modifies the samples to model a form of distortion, and outputs the modified samples. The output of the AGM is provided to an acoustic echo estimation (AEE) module that uses adaptive algorithms to compensate for substantially linear changes in the echo characteristics of the environment in which the loudspeaker and microphone are located.

CROSS-REFERENCE TO RELATED APPLICATION

This application is a continuation of U.S. Ser. No. 09/699,188, filed onOct. 27, 2000, entitled DISTORTION COMPENSATION IN AN ACOUSTIC ECHOCANCELER, by inventor James H. Parry, currently pending [Attorney DocketNo. 38715-P014US].

BACKGROUND OF THE INVENTION

1. Field of Invention

This invention pertains in general to telephony and televideoconferencing and in particular to performing acoustic echo cancellationon potentially distorted audio signals.

2. Background of Invention

Two-way audio communications systems, such as speakerphones and videocommunications systems having audio capabilities, utilize both amicrophone and a loudspeaker. The microphone transmits speech and othersounds from the local terminal to remote terminals while the loudspeakeremits sounds received from remote terminals. In a typical hands-freesystem, the loudspeaker and microphone are located in close proximityand sounds produced by the loudspeaker are picked up by the microphone.Without signal processing, therefore, a feedback loop is easily createdbetween the loudspeaker and microphone. This feedback can cause theloudspeaker to emit an undesirable “howling” noise and cause the remoteterminals to hear echoes.

One simple technique for eliminating feedback is to provide half-duplexswitching where only the microphone or the loudspeaker is active at anygiven instant. In a typical half-duplex system, the loudspeaker isactive until a sound is detected at the microphone. Then, theloudspeaker becomes inactive and the microphone becomes active for theduration of the sound. Half-duplex systems have many inherent problems,not the least of which is that a slight noise may unintentionally causethe loudspeaker to cut out. As a result, it is often difficult toconduct a normal conversation with a system using half-duplex switching.

More sophisticated audio communications systems use acoustic echocancellation (AEC) to reduce echoes and eliminate howling. An AEC systemtypically utilizes a sample-by-sample copy of the signal going to theloudspeaker as the basis for an estimate of the echo returning throughthe microphone, as taught in U.S. Pat. No. 4,965,822, entitled FULLDUPLEX SPEAKERPHONE, which issued on Oct. 23, 1990 and is incorporatedby reference herein. This estimated echo is subtracted on asample-by-sample basis in an attempt to separate out only that portionof the microphone signal due to sounds coming from sources other thanthe speaker. An adaptive AEC uses a filter having slowly adjustedweights to form the echo estimate in an effort to more accuratelysubtract the echo from the returned audio signal. Subsequentconditioning performed on the output of the AEC may include automaticgain control (AGC) and perceived noise reduction.

A problem with the above approach is that the loudspeakers do notproduce sound pressure signals that are exactly proportional to thedriving voltage (or current). Likewise, microphones are imperfect in ananalogous sense. There may also be other sources of distortion withinthe sound system, such as amplifiers, analog-to-digital (A/D) anddigital-to-analog (D/A) converters, and perhaps even the user'senvironment. Existing AEC systems do not accurately remove the nonlinearcomponents of the returned signal due to these sources of potentialdistortion. As a result, a badly distorted form of the echo can passthrough the echo cancellation process. Another undesirable effect ofthese introduced distortions is that the adaptation of the AECparameters is degraded, leading to a greater perceived echo.

One potential solution to the problem of degraded AEC adaptation is touse a reduced adaptation rate during periods of very loud sound output.This technique is used, for example, in U.S. Pat. No. 6,650,701,entitled APPARATUS AND METHOD FOR CONTROLLING AN ACOUSTIC ECHO CANCELER,which issued on Nov. 18, 2003, and incorporated by reference herein.However, reducing the adaptation rate has the undesirable effect ofslowing the system's response to a changing acoustic environment such aswhen users are in motion and/or the room temperature fluctuates.

Another potential solution is to use higher quality loudspeakers andother components. This solution, however, carries with it considerableexpense and places severe limitations on the designs of the equipment.High-quality loudspeakers are typically large and heavy and generatestrong external magnetic fields. Often, the audio communications systemis integrated into another sound system, such as the audio subsystem ofa laptop computer, where a high-quality loudspeaker cannot be used.

Therefore, there is a need for a technique for more accuratelyestimating the echo when performing acoustic echo cancellation. There isalso a need for a technique for more accurately adapting the estimatedecho in response to changing acoustic characteristics.

DISCLOSURE OF THE INVENTION

The above needs are met by using modules to estimate the nonlineardistortions in the audio signal returned from the microphone that wereintroduced by the loudspeaker, microphone, and related components.

A typical audio communications system has a plurality of terminalscoupled to a switch. The terminals can include, for example, dedicatedspeakerphones, desktop handsets with or without speakerphonecapabilities, cellular phones, and/or personal computer (PC) systemswith audio capabilities. The switch may be dedicated to audiocommunications, as is a private branch exchange (PBX), or distributedand multifunctional, as is an Internet server.

Each terminal preferably includes a microphone and a loudspeaker. Anamplifier amplifies the electrical signals produced by the microphoneand provides its output to an A/D converter. The A/D converter outputsequivalent digital samples. The loudspeaker is driven by anotheramplifier which, in turn, is driven by the output of a D/A converter.The D/A converter receives digital samples representing the soundpressure waves to be produced by the loudspeaker.

In order to cancel echoes of the loudspeaker picked up by themicrophone, the audio communications system has an acoustic echocancellation (AEC) module. The AEC module can be located in the terminalor elsewhere in the audio communications system. U.S. patent applicationSer. No. 11/124,772, entitled COMMUNICATIONS SYSTEM AND METHOD UTILIZINGCENTRALIZED SIGNAL PROCESSING, filed on May 9, 2005, and incorporated byreference herein, describes potential locations of the AEC module. TheAEC module preferably receives the digital signal sent to theloudspeaker and the digital signal received from the microphone.

The digital loudspeaker signal is processed by an audio generationmodule (AGM) to model the substantially nonlinear distortions that canoccur during the process of playing the audio signal at the loudspeaker.The AGM includes a modeling path comprised of one or more distortionmodules. Each distortion module receives digital samples as input,modifies the samples to model a form of distortion, and outputs themodified samples. A distortion module can be adaptive or it can bepartly or wholly pre-established. Preferably, the AGM can add or removedistortion modules from the modeling path at any time in response tocharacteristics of the digital samples or under direction from othermodules. Distortions that can be modeled by the distortion modules inthe AGM modeling path include, for example, amplifier clipping,loudspeaker voice coil displacement, harmonic distortion introduced bythe loudspeaker, and hysterisis in an iron-core inductor.

The AGM outputs digital sample values to an acoustic echo estimation(AEE) module. The AEE module preferably uses known adaptive algorithmsto adapt the digital samples to compensate for substantially linearchanges in the echo characteristics of the environment in which theloudspeaker and microphone are located. For example, the AEE module canmodify the digital samples to account for changes in echo attenuationdue to relocation of people in the vicinity of the microphone.

The output of the AEE module is received by an audio sensing module(ASM). The ASM performs a function similar to the AGM, except that theASM models substantially nonlinear distortions that occur while sensingthe audio signal. Accordingly, the ASM models distortions such asmicrophone 116 centerclipping, amplifier zero crossing distortion,saturation in either the microphone or the amplifier, and distortionsintroduced by the A/D converter. The output of the ASM represents theestimate of the echo of the loudspeaker signal in the signal receivedfrom the microphone.

The digital samples returned from the microphone and the output of theASM are received by an adder module. The adder module subtracts theestimated echo received from the ASM from the samples returned from themicrophone, thereby removing at least part of the estimated echo of theloudspeaker from the microphone signal.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a high-level block diagram of an audio communications systemaccording to an embodiment of the present invention;

FIG. 2 is a block diagram illustrating various components of the audiocommunications system including an acoustic echo cancellation (AEC)module; and

FIG. 3 is a lower-level view of an exemplary audio generation module inthe AEC module.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

FIG. 1 is a high-level block diagram of an audio communications system100 according to an embodiment of the present invention. A plurality ofterminals 110A-D are coupled to a switch 112 via communications links114A-D. The terminal types can be heterogeneous or homogeneous. In oneembodiment, the terminals include: dedicated speakerphones, desktophandsets with or without speakerphone capabilities, cellular phones,and/or personal computer (PC) systems with audio capabilities. As usedherein, the phrase “audio communications system” also includes videoconferencing systems having audio capabilities. Each terminal 110, ofwhich terminal 110A is representative, preferably includes a microphone116A and a loudspeaker 118A. As is known in the art, the microphone 116converts sound pressure waves into electrical signals and theloudspeaker 118 converts electrical signals into sound pressure waves.

The communications links 114 carry audio data representative of soundspicked up by the microphone 116 and to be played by the loudspeaker 118to/from the switch 112. The communications links 114 may be wired orwireless. Moreover, the links 114 may include dedicated private links,shared links utilizing a publicly-accessible telephone network, and/orlinks using a public or private data communications network such as theInternet. Data traveling over the links 114 may pass through one or moreswitches or link types before reaching the switch 112 or terminal 110,although a preferred embodiment of the present invention treats a linkpassing through multiple links and switches as a single logical link.The data carried by the communications links 114 can be digital and/oranalog. If the data is digital, it is preferably transmitted as a seriesof discrete data packets, such as Internet protocol (IP) packets. In oneembodiment, the digital data is encoded into a compressed format.

The switch 112 switches and routes communications among the terminals.The switch 112 can be, for example, a private branch exchange (PBX)located at a business or other entity, a publicly-accessible switchoperated by a telephone company or other entity providing audiocommunications, or an Internet server supporting Internet telephony.Thus, the term “switch” includes any device or combination of devicescapable of providing the switching and other functionality attributed tothe switch herein.

In one embodiment, the terminals 110 and/or switch 112 have one or moreof the components found in a typical computer system, including aprocessing unit, random access memory (RAM), read-only memory (ROM), astorage device such as a hard drive, and/or other hardware and softwarefor providing the functionality described herein. Aggregations ofmachine-executable code, data, circuitry, and/or data storage areas forperforming a specific purpose or purposes are referred to as “modules.”Different modules may share common code, data, and/or circuitry. Themodules include, for example, signal processing modules,digital-to-analog (D/A) and analog-to-digital (A/D) converter modules,and amplifier modules. Modules may hold in their storage areas previousvalues of signals and current statistics derived therefrom. Modules canalso use adaptive techniques, or training, to perform the modules'functionalities. As used herein, the terms “adaptation” and “training”are interchangeable and refer to acting on a signal responsive toprevious values of that signal or other signals, statistics derived fromthe signals, and/or external controls or sensors.

FIG. 2 is a block diagram illustrating various components of the audiocommunications system including an acoustic echo cancellation (AEC)module 210. FIG. 2 illustrates the microphone 116 of FIG. 1 having itsoutput coupled to an amplifier 212. As is known in the art, themicrophone 116 converts sound pressure waves into electrical signals.The amplifier 212 amplifies the electrical signals and provides itsoutput to an A/D converter 214. The A/D converter 214 outputs digitalsample values representative of the sound pressure waves to the AECmodule 210.

FIG. 2 also illustrates the speaker 118 of FIG. 1. The speaker 118generates sound pressure waves in response to an input received from anamplifier 216. The amplifier 216, in turn, is driven by the output of aD/A converter 218. The D/A converter 218 receives digital sample valuesrepresenting the sound pressure waves as input from the link 114 oranother source.

In general, the AEC module 210 estimates the echo of sounds played bythe loudspeaker 118 that are picked up by the microphone 116, subtractsthe estimated echo from the microphone's audio signal, and outputs theresulting echo-cancelled signal. In one embodiment, the AEC module 210is located in the terminal 110. Accordingly, the output of the AECmodule 210 is passed over the communications links 114 to the switch112. In alternative embodiments, the AEC module 210 is located in theswitch 112 or anywhere else that echo cancellation is desired andrepresentations of the loudspeaker and microphone signals are available.

Turning to the AEC module 210 itself, the digital samples representingthe audio signal sensed by the microphone output by the A/D converter214 are received by an adder module 220. The adder module 220 alsoreceives an input 222 providing digital samples representing the echofrom the loudspeaker 118 estimated to be present in the microphonesignal. The adder module 220 preferably adds the negative of theestimated echo to the signal received from the A/D converter 214.Preferably, the adder module 220 works on a sample-by-sample basis. Inone embodiment, both the estimated echo samples received from the input222 and the sample values received from the A/D converter 214 bearsequencing information that the adder module 220 uses to match thesamples. U.S. patent application Ser. No. 09/660,205, incorporated byreference herein, discloses additional details related to the sequencinginformation.

The output of the adder module 220 is passed to a perceived noisereduction module 224. This module 224 preferable reduces perceived noisein the audio signal. Techniques for reducing perceived noise are wellknown in the art.

The output of the perceived noise reduction module is preferably passedto an automatic gain control (AGC) module 226. As is known in the art,the AGC module 226 preferably isolates times during which local speechis thought to be present in the input signal and adjusts the signal gainso that the speech is near a predetermined level when considered onaverage. The AGC module 226 can use adaptive techniques to perform AGC.The output 228 of the AGC module 226 is preferably provided to theswitch 112 via the communications links 114 as described above.

The AEC module 210 also receives an input 230 carrying digital samplevalues representing the audio signal being sent to the loudspeaker 118of the terminal 110. If the AEC module 210 is located in the terminal110, then this input 230 is received from the switch 112 via thecommunications links 114. The loudspeaker 118 digital sample values arereceived by an audio generation module (AGM) 232 within the AEC module210.

The AGM 232 preferably modifies the digital sample values to modelsubstantially nonlinear distortions that can occur during the process ofgenerating the audio signal. FIG. 3 is a block diagram illustrating alower-level view of the AGM 232 according to an exemplary embodiment ofthe present invention. The AGM 232 includes a modeling path 310comprised of logical interconnects among one or more distortion modules312 that operate on the digital samples traveling through the path. Eachdistortion module 312 receives digital samples as input, modifies thesamples to model a form of distortion, and outputs the modified samples.Preferably, the AGM 232 can add or remove distortion modules from themodeling path 310 at any time in response to characteristics of thedigital samples or under direction from other modules.

The AGM 232 preferably models effects which are substantially nonlinear.Certain embodiments utilize artificial neural networks (ANNs) to achieveadaptation. Those ANNs which are not adaptive may be present at the timeof manufacture and do not require feedback for further adaptation. ANNsin adaptive modules 312 utilize internal and/or external feedback. Suchfeedback may be from other distortion modules 312, from the loudspeakerdigital signal, and/or from the microphone signal before or after theadder module 220. These many possible feedback paths have been omittedfrom the modeling path 310 in FIG. 3 in order to clarify the teachingsof the present invention.

The example of a modeling path 310 illustrated in the AGM 232 of FIG. 3has three distortion modules 312A, 312B, 312C arranged in sequence. Eachdistortion module 312 preferably contains a filter or other operatorthat acts on the input samples. The module 312 can be adaptive or it canbe partly or wholly pre-established. Likewise, the module 312 canoperate in the time or frequency domains. The module 312 can also act inresponse to short or long-term signal characteristics to model effectssuch as heat build-up.

Preferably, each distortion module 312 independently models a form ofdistortion. In FIG. 3, the first distortion module 312A models amplifierclipping by enforcing a hard limit on signal amplitudes. Thus, thisdistortion module 312A models the effects of the speaker amplifier 216in the terminal 110 on the analog signal sent to the loudspeaker 118.The second distortion module 312B models loudspeaker 118 voice coil (orequivalent structure) displacement. In one embodiment, this distortionmodule 312B estimates the nonlinear relationship between the voice coildisplacement and the driving current. In one embodiment, the drivingcurrent estimate received by the voice coil displacement module 312B isgenerated by the amplifier clipping module 312A and so may be anonlinear representation of the loudspeaker digital samples. The thirddistortion module 312C models harmonic distortion introduced by theloudspeaker 118. In one embodiment, this distortion module 312C appliesharmonic distortion with a strength modulated by the energy in thespectral components subject to the distortion. Thus, this distortionmodule 312C mimics the operation of a loudspeaker 118 driven with highelectrical amplitudes where diaphragms distort and resonate. Otherdistortion modules 312 that may be utilized in the modeling path 310include modules that account for distortions introduced by the D/Aconversion module 218 and modules that account for hysterisis in ironcore inductors.

In one embodiment of the present invention, the distortion modules 312are tailored to model the distortions introduced by specific types ofhardware. For example, if the AGM 232 is located in the terminal 110,the amplifier clipping 310A and voice coil displacement 310B modules canbe specifically tailored for the amplifiers and voice coils included inthe terminal 110.

The AGM 232 outputs digital sample values representing the distortedaudio signal to an acoustic echo estimation (AEE) module 234. The AEEmodule 234 preferably uses adaptive algorithms to adapt the digitalsamples to compensate for substantially linear changes in the echocharacteristics of the environment in which the loudspeaker 118 andmicrophone 116 are located. For example, the AEE module 234 can modifythe digital samples to account for changes in echo attenuation due torelocation of people in the vicinity of the microphone 116.

The digital sample values output by the AEE module 234 are preferablyreceived by an audio sensing module (ASM) 236. The ASM 236 preferablymodifies the digital sample values to model distortions that can occurin the process of sensing the audio signal. Like the AGM 232, the ASM236 preferably includes a modeling path comprised of logicalinterconnects among one or more distortion modules. The modeling pathfor the ASM 236 is not shown in the figures because it would beredundant in view of FIG. 3. Also like the AGM 232, the ASM 236preferably models substantially nonlinear distortions. Unlike the AGM232, the ASM preferably models distortions such as microphone 116centerclipping, amplifier zero crossing distortion, saturation in eitherthe microphone or the amplifier, and/or distortions introduced by theA/D converter 214. The output of the ASM 236 is provided to the addermodule 220 and becomes the input signal representing the echo from theloudspeaker 118 estimated to be present in the microphone signaldescribed above.

Accordingly, the AEC module 210 of the present invention accuratelymodels the effects of distortion on the audio signals. The modeled typesof distortion include nonlinear distortions introduced while generatingand sensing the audio signal and linear echoes introduced responsive toroom characteristics. This distortion modeling enables the AEC to moreaccurately cancel the echo in the signal received from the microphone116.

The above description is included to illustrate the operation of thepreferred embodiments and is not meant to limit the scope of theinvention. The scope of the invention is to be limited only by thefollowing claims. From the above discussion, many variations will beapparent to one skilled in the relevant art that would yet beencompassed by the spirit and scope of the invention.

1. A module in an audio communication system, comprising: a first inputfor receiving a first audio signal; a second input for receiving asecond audio signal, wherein at least a portion of the second audiosignal is an echo of the first audio signal, the echo having adistortion; a plurality of aggregations, the aggregations implementingadaptive techniques wherein the aggregations comprise: a distortionmodule receiving the first audio signal, the distortion module adaptedto model the distortion of the first audio signal and produced adistorted signal; an audio sensing module receiving the first audiosignal and adapted to use the distortion module to model a distortionthat occurs responsive to sensing the second audio signal; and an addermodule for receiving the distorted signal and the second audio signaland adapted to use the distorted signal to remove at least part of theecho from the second audio signal.
 2. The module of claim 1, wherein thefirst and second audio signals bear sequencing information and whereinthe adder module is adapted to use the sequencing information to removeat least part of the echo from the second audio signal.
 3. The module ofclaim 1, further comprising: an audio generation module receiving thefirst audio signal and adapted to use the distortion module to model adistortion that occurs responsive to playing the first audio signalthrough a loudspeaker.
 4. The module of claim 3, wherein the audiogeneration module comprises: a modeling path having one or moredistortion modules that model distortions on the first audio signal. 5.The module of claim 4, wherein each distortion module models a differenttype of distortion.
 6. The module of claim 4, wherein the audiogeneration module alters the modeling path in real-time responsive todistortions that may occur on the first audio signal.
 7. The module ofclaim 3, wherein the distortion module models an effect of amplifierclipping on the first audio signal.
 8. The module of claim 3, whereinthe distortion module models an effect of voice coil displacement onsound pressure waves produced by the loudspeaker responsive to the firstaudio signal.
 9. The module of claim 3, wherein the distortion modulemodels an effect of hysterisis in an iron core inductor on the firstaudio signal.
 10. The module of claim 3, wherein the distortion modulemodels an effect of harmonic distortion on sound pressure waves producedby the loudspeaker responsive to the first audio signal.
 11. The moduleof claim 1, further comprising: an acoustic echo estimation modulereceiving the first audio signal and for adapting the first audio signalto compensate for substantially linear changes in the second audiosignal.
 12. The module of claim 1, further comprising: an audio sensingmodule receiving the first audio signal and adapted to use thedistortion module to model a distortion that occurs responsive tosensing the second audio signal.
 13. The module of claim 12, wherein theaudio sensing module comprises: a modeling path having one or moredistortion modules that model distortions on the second audio signal.14. The module of claim 13, wherein each distortion module models adifferent type of distortion.
 15. The module of claim 13, wherein theaudio sensing module alters the modeling path in real-time responsive todistortions that may occur on the second audio signal.
 16. The module ofclaim 12, wherein the distortion module models an effect of microphonecenterclipping on the second audio signal.
 17. The module of claim 12,wherein the distortion module models an effect of amplifier zerocrossing distortion on the second audio signal.
 18. The module of claim1, wherein the distortion module models a pre-established distortion.19. The module of claim 1, wherein the distortion module is adaptive.20. The module of claim 1, wherein the distortion module models anonlinear distortion.
 21. The module of claim 1, wherein the distortionmodule operates in a frequency domain.
 22. A method of canceling an echoin an audio signal, comprising the steps of: providing a plurality ofcommunication links for carrying data, wherein the data comprises analogor digital data; receiving a first audio signal; receiving a secondaudio signal, wherein at least a portion of the second audio signal is adistorted audio signal; encoding any digital data received into acompressed format; modeling one or more types of distortions on thefirst audio signal to produce a distorted audio signal; and subtractingthe distorted audio signal from the second audio signal to at leastpartially cancel the distorted echo of the first audio signal from thesecond audio signal; wherein the modeling path models distortions thatoccur responsive to sensing the second audio signal.
 23. The method ofclaim 22, wherein the modeling step comprises the step of: adaptivelymodeling one or more types of distortion.
 24. The method of claim 22,wherein the modeling step comprises the step of: modeling apre-established type of distortion.
 25. The method of claim 22, furthercomprising the step of: retrieving sequencing information from the firstand second audio signals; wherein the subtracting step uses thesequencing information to at least partially cancel the distorted echoof the first audio signal from the second audio signal.
 26. The methodof claim 22, wherein the modeling step comprises the step of: passingthe first audio signal through a modeling path comprising one or moredistortion modules, each distortion module applying a type of distortionto the first audio signal.
 27. The method of claim 26, wherein themodeling path models distortions that occur responsive to playing thefirst audio signal through a loudspeaker.
 28. The method of claim 27,wherein the passing step comprises the step of: passing the first audiosignal through a distortion module that models an effect of amplifierclipping on the first audio signal.
 29. The method of claim 27, whereinthe passing step comprises the step of: passing the first audio signalthrough a distortion module that models an effect of voice coildisplacement on sound pressure waves produced by the loudspeakerresponsive to the first audio signal.
 30. The method of claim 27,wherein the passing step comprises the step of: passing the first audiosignal through a distortion module that models an effect of harmonicdistortion on the sound pressure waves produced by the loudspeakerresponsive to the first audio signal.
 31. The method of claim 27,wherein the passing step comprises the step of: passing the first audiosignal through a distortion module that models an effect of hysterisisin inductors containing iron on the first audio signal.
 32. The methodof claim 26, wherein the modeling path models distortions that occurresponsive to sensing the second audio signal.
 33. The method of claim32, wherein the passing step comprises the step of: passing the firstaudio signal through a distortion module that models an effect ofmicrophone centerclipping on the second audio signal.
 34. The module ofclaim 32, wherein the passing step comprises the step of: passing thefirst audio signal through a distortion module that models an effect ofamplifier zero crossing distortion on the second audio signal.
 35. Aterminal for an audio communications system, the terminal comprising: aplurality of communication links, wherein the links include wired orwireless links comprising dedicated private links, shared links and/oruse of a public or private data communication network; a loudspeaker forproducing sound pressure waves carried via the communication links, theloudspeaker responsive to a received first signal; a microphone forconverting sound pressure waves into a second signal, wherein a portionof the second signal represents an echo of the sound pressure wavesproduced by the loudspeaker; a distortion module receiving the firstsignal and adapted to modify the first signal to model a type ofdistortion to produce a distorted signal; an adder module for removingat least a portion of the echo of the sound pressure waves produced bythe loudspeaker from the second signal responsive to the distortedsignal; and an audio sensing module receiving the first signal andadapted to use the distortion module to model a distortion that occursresponsive to sensing the audio signal wherein the audio sensing modulehas a modeling path comprising one or more distortion modules that modeldistortions on the second signal.
 36. The terminal of claim 35, whereinthe first and second signals bear sequencing information and wherein theecho cancellation module is adapted to use the sequencing information toremove at least part of the echo from the second signal.
 37. Theterminal of claim 35, further comprising: an audio generation modulereceiving the first signal and adapted to use the distortion module tomodel a distortion that occurs responsive to playing the first signalthrough the loudspeaker.
 38. The terminal of claim 37, wherein the audiogeneration module has a modeling path comprising one or more distortionmodules that model distortions on the first signal.
 39. The terminal ofclaim 38, wherein each of the one or more distortion modules models adifferent type of distortion.
 40. The terminal of claim 38, wherein theaudio generation module alters the modeling path in real-time responsiveto distortions that may occur on the first signal.
 41. The terminal ofclaim 37, wherein the distortion module models an effect of amplifierclipping on the first signal.
 42. The terminal of claim 37, wherein thedistortion module models an effect of voice coil displacement on thesound pressure waves produced by the loudspeaker.
 43. The terminal ofclaim 37, wherein the distortion module models an effect of hysterisisin an iron core inductor on the first signal.
 44. The terminal of claim37, wherein the distortion module models an effect of harmonicdistortion introduced by the loudspeaker on the sound pressure waves.45. The terminal of claim 35, further comprising: an acoustic echoestimation module receiving the first signal and for adapting the firstsignal to compensate for substantially linear changes in the secondsignal.
 46. The terminal of claim 35, further comprising: an audiosensing module receiving the first signal and adapted to use thedistortion module to model a distortion that occurs responsive tosensing the audio signal.
 47. The terminal of claim 46, wherein theaudio sensing module has a modeling path comprising one or moredistortion modules that model distortions on the second signal.
 48. Theterminal of claim 47, wherein each of the one or more distortion modulesmodels a different type of distortion.
 49. The terminal of claim 47,wherein the audio sensing module alters the modeling path in real-timeresponsive to distortions that may occur on the second audio signal. 50.The terminal of claim 46, wherein the distortion module models an effectof microphone centerclipping on the second signal.
 51. The terminal ofclaim 46, wherein the distortion module models an effect of amplifierzero crossing distortion on the second signal.
 52. A system for audiocommunications, the system comprising: a plurality of communicationlinks, wherein the links include wired or wireless links comprisingdedicated private links, shared links and/or use of a public or privatedata communications network; a loudspeaker operable to produce soundpressure waves carried via the communication links, the loudspeakerresponsive to a first signal; a microphone operable to convert soundpressure waves into a second signal, the microphone positioned to sensesound pressure waves from the loudspeaker, the microphone havingnon-linear characteristics that introduce distortion to the secondsignal; a distortion module interfaced with the first signal andoperable to generate a distorted signal modeled to the distortionintroduced by the microphone in the sensing of the loudspeaker soundpressure waves; and an adder module interfaced with the second signaland the distorted signal, the adder operable to remove the distortedsignal from the second signal.